Over the past several years there has been, and continues to be, a tremendous amount of activity in the area of efficient encoding of speech. For an evolving digital telephone network, a most important application is the possible replacement of the 64,000 bit-per-second (b/s) PCM signal (8 bits per time slot, repeated at an 8 kHz rate) with other coding algorithms for telephony--both in the public switched and private line networks. The reason, of course, is to achieve bandwidth compression.
For a realistic mix of input speech and voiceband data, adaptive differential PCM appears to be a most promising approach. One form of adaptive differential PCM coding is disclosed, for example, in copending application Ser. No. 343,355, filed Jan. 27, 1982 now U.S. Pat. No. 4,437,087 issued Mar. 13, 1984 and can be considered a benchmark since a single encoding with this ADPCM coder at 32 kb/s is near to being subjectively equivalent to 64 kb/s .mu.255 PCM. In some communications networks, however, it is possible to encounter multiple conversions from 64 kb/s PCM-to-32 kb/s ADPCM-to-64 kb/s-PCM. In these applications, it is important to prevent accumulation of distortion in the multiple PCM-to-ADPCM-to-PCM code conversions because of noise caused by quantizing the digital signals and noise caused by truncating values of digital signals used in the code converters.
An attempt towards this goal is described in an article by Hideyo Murakami entitled, "A Low Noise ADPCM-Log PCM Code Converter", 1979 International Symposium on Circuits and Systems Proceedings, IEEE catalog No. 79 CH1421-7 CAS, pages 969-970. However, the disclosed arrangement assumes a uniform ADPCM quantizer characteristic and distortion could result in those applications in which a non-uniform quantizer characteristic is employed. Additionally, the truncation noise problem is apparently not addressed.
More recently, possible distortion accumulation because of multiple code conversions is prevented in an ADPCM codec by controllably modifying the ADPCM code word as disclosed in copending application Ser. No. 435,968, filed Oct. 22, 1982 now U.S. Pat. No. 4,437,087. One problem with the disclosed arrangement is that one portion of the distortion prevention accumulation algorithm is included in the PCM-to-ADPCM coder while another portion is included in the ADPCM-to-PCM decoder. Another problem is that possible distortion caused by truncating digital signals, i.e., eliminating factional portions of digital signals used in processing is not addressed directly. In this prior arrangement, certain values of the ADPCM quantizer scale factors have been excluded from use in an attempt to eliminate noise caused by truncating the digital signals. Unfortunately, the ADPCM quantizer scale factor values to be excluded must be found experimentally, which can be a time consuming and tedious procedure. Moreover, it is still not known if distortion will result under certain signal conditions.
Thus, although the prior known arrangements may function satisfactorily in certain applications, they may not function satisfactorily in others.